[gstreamer-bugs] [Bug 549774] audio / video synchronization glitch during rtsp playback

[GStreamer (bugzilla.gnome.org)]

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  GStreamer | gst-plugins-base | Ver: 0.10.18

Wim Taymans changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
             Status|UNCONFIRMED                 |NEEDINFO




------- Comment #11 from Wim Taymans  2008-09-05 13:56 UTC -------
I reworked the code a bit to better handle this case, had to move some things
around resulting in this commit: If you can test cvs, please report, if you can
provide the problematic clip, I'll test for you.

        * gst/rtpmanager/gstrtpbin.c: (on_sender_timeout),
        (create_session), (gst_rtp_bin_associate),
        (gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init),
        (gst_rtp_bin_request_new_pad):
        * gst/rtpmanager/gstrtpbin.h:
        Add signal to notify listeners when a sender becomes a receiver.
        Tweak lip-sync code, don't store our own copy of the ts-offset of the
        jitterbuffer, don't adjust sync if the change is less than 4msec.
        Get the RTP timestamp <-> GStreamer timestamp relation directly from
        the jitterbuffer instead of our inaccurate version from the source.

        * gst/rtpmanager/gstrtpjitterbuffer.c:
        (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
        (gst_rtp_jitter_buffer_get_sync):
        * gst/rtpmanager/gstrtpjitterbuffer.h:
        Add G_LIKELY macros, use global defines for max packet reorder and
        dropouts.
        Reset the jitterbuffer clock skew detection when packets seqnums are
        changed unexpectedly.

        * gst/rtpmanager/gstrtpsession.c: (on_sender_timeout),
        (gst_rtp_session_class_init), (gst_rtp_session_init):
        * gst/rtpmanager/gstrtpsession.h:
        Add sender timeout signal.

        * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
        (calculate_skew), (rtp_jitter_buffer_insert),
        (rtp_jitter_buffer_get_sync):
        * gst/rtpmanager/rtpjitterbuffer.h:
        Add some G_LIKELY macros.
        Keep track of the extended RTP timestamp so that we can report the RTP
        timestamp <-> GStreamer timestamp relation for lip-sync.
        Remove server timestamp gap detection code, the server can sometimes
        make a huge gap in timestamps (talk spurts,...) see #549774.
        Detect timetamp weirdness instead by observing the sender/receiver
        timestamp relation and resync if it changes more than 1 second.
        Add method to report about the current rtp <-> gst timestamp relation
        which is needed for lip-sync.

        * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
        (on_sender_timeout), (check_collision), (rtp_session_process_sr),
        (session_cleanup):
        * gst/rtpmanager/rtpsession.h:
        Add sender timeout signal.
        Remove inaccurate rtp <-> gst timestamp relation code, the
        jitterbuffer can now do an accurate reporting about this.

        * gst/rtpmanager/rtpsource.c: (rtp_source_init),
        (rtp_source_update_caps), (calculate_jitter),
        (rtp_source_process_rtp):
        * gst/rtpmanager/rtpsource.h:
        Remove inaccurate rtp <-> gst timestamp relation code.

        * gst/rtpmanager/rtpstats.h:
        Define global max-reorder and max-dropout constants for use in various
        subsystems.


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You can add comments to this bug at http://bugzilla.gnome.org/show_bug.cgi?id=549774.